summaryrefslogtreecommitdiffstats
path: root/apps/plugins/mpegplayer/audio_thread.c
blob: f976fd600761e743453902e2ab93a4e18002cb13 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
/***************************************************************************
 *             __________               __   ___.
 *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
 *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
 *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
 *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
 *                     \/            \/     \/    \/            \/
 * $Id$
 *
 * mpegplayer audio thread implementation
 *
 * Copyright (c) 2007 Michael Sevakis
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
 * as published by the Free Software Foundation; either version 2
 * of the License, or (at your option) any later version.
 *
 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
 * KIND, either express or implied.
 *
 ****************************************************************************/
#include "plugin.h"
#include "mpegplayer.h"
#include "codecs/libmad/bit.h"
#include "codecs/libmad/mad.h"

/** Audio stream and thread **/
struct pts_queue_slot;
struct audio_thread_data
{
    struct queue_event ev;  /* Our event queue to receive commands */
    int state;              /* Thread state */
    int status;             /* Media status (STREAM_PLAYING, etc.) */
    int mad_errors;         /* A count of the errors in each frame */
    unsigned samplerate;    /* Current stream sample rate */
    int nchannels;          /* Number of audio channels */
    struct dsp_config *dsp; /* The DSP we're using */
};

/* The audio thread is stolen from the core codec thread */
static struct event_queue audio_str_queue SHAREDBSS_ATTR;
static struct queue_sender_list audio_str_queue_send SHAREDBSS_ATTR;
struct stream audio_str IBSS_ATTR;

/* libmad related definitions */
static struct mad_stream stream IBSS_ATTR;
static struct mad_frame  frame IBSS_ATTR;
static struct mad_synth  synth IBSS_ATTR;

/*sbsample buffer for mad_frame*/
mad_fixed_t sbsample[2][36][32];

/* 2567 bytes */
static unsigned char mad_main_data[MAD_BUFFER_MDLEN];

/* There isn't enough room for this in IRAM on PortalPlayer, but there
   is for Coldfire. */

/* 4608 bytes */
#if defined(CPU_COLDFIRE) || defined(CPU_S5L870X)
static mad_fixed_t mad_frame_overlap[2][32][18] IBSS_ATTR;
#else
static mad_fixed_t mad_frame_overlap[2][32][18];
#endif

/** A queue for saving needed information about MPEG audio packets **/
#define AUDIODESC_QUEUE_LEN  (1 << 5) /* 32 should be way more than sufficient -
                                         if not, the case is handled */
#define AUDIODESC_QUEUE_MASK (AUDIODESC_QUEUE_LEN-1)
struct audio_frame_desc
{
    uint32_t time;  /* Time stamp for packet in audio ticks       */
    ssize_t  size;  /* Number of unprocessed bytes left in packet */
};

 /* This starts out wr == rd but will never be emptied to zero during
    streaming again in order to support initializing the first packet's
    timestamp without a special case */
struct
{
    /* Compressed audio data */
    uint8_t *start;  /* Start of encoded audio buffer */
    uint8_t *ptr;    /* Pointer to next encoded audio data */
    ssize_t used;    /* Number of bytes in MPEG audio buffer */
    /* Compressed audio data descriptors */
    unsigned read, write;
    struct audio_frame_desc *curr; /* Current slot */
    struct audio_frame_desc descs[AUDIODESC_QUEUE_LEN];
} audio_queue;

static inline int audiodesc_queue_count(void)
{
    return audio_queue.write - audio_queue.read;
}

static inline bool audiodesc_queue_full(void)
{
    return audio_queue.used >= MPA_MAX_FRAME_SIZE + MAD_BUFFER_GUARD ||
            audiodesc_queue_count() >= AUDIODESC_QUEUE_LEN;
}

/* Increments the queue tail postion - should be used to preincrement */
static inline void audiodesc_queue_add_tail(void)
{
    if (audiodesc_queue_full())
    {
        DEBUGF("audiodesc_queue_add_tail: audiodesc queue full!\n");
        return;
    }

    audio_queue.write++;
}

/* Increments the queue head position - leaves one slot as current */
static inline bool audiodesc_queue_remove_head(void)
{
    if (audio_queue.write == audio_queue.read)
        return false;

    audio_queue.read++;
    return true;
}

/* Returns the "tail" at the index just behind the write index */
static inline struct audio_frame_desc * audiodesc_queue_tail(void)
{
    return &audio_queue.descs[(audio_queue.write - 1) & AUDIODESC_QUEUE_MASK];
}

/* Returns a pointer to the current head */
static inline struct audio_frame_desc * audiodesc_queue_head(void)
{
    return &audio_queue.descs[audio_queue.read & AUDIODESC_QUEUE_MASK];
}

/* Resets the pts queue - call when starting and seeking */
static void audio_queue_reset(void)
{
    audio_queue.ptr = audio_queue.start;
    audio_queue.used = 0;
    audio_queue.read = 0;
    audio_queue.write = 0;
    rb->memset(audio_queue.descs, 0, sizeof (audio_queue.descs));
    audio_queue.curr = audiodesc_queue_head();
}

static void audio_queue_advance_pos(ssize_t len)
{
    audio_queue.ptr        += len;
    audio_queue.used       -= len;
    audio_queue.curr->size -= len;
}

static int audio_buffer(struct stream *str, enum stream_parse_mode type)
{
    int ret = STREAM_OK;

    /* Carry any overshoot to the next size since we're technically
       -size bytes into it already. If size is negative an audio
       frame was split across packets. Old has to be saved before
       moving the head. */
    if (audio_queue.curr->size <= 0 && audiodesc_queue_remove_head())
    {
        struct audio_frame_desc *old = audio_queue.curr;
        audio_queue.curr = audiodesc_queue_head();
        audio_queue.curr->size += old->size;
        old->size = 0;
    }

    /* Add packets to compressed audio buffer until it's full or the
     * timestamp queue is full - whichever happens first */
    while (!audiodesc_queue_full())
    {
        ret = parser_get_next_data(str, type);
        struct audio_frame_desc *curr;
        ssize_t len;

        if (ret != STREAM_OK)
            break;

        /* Get data from next audio packet */
        len = str->curr_packet_end - str->curr_packet;

        if (str->pkt_flags & PKT_HAS_TS)
        {
            audiodesc_queue_add_tail();
            curr = audiodesc_queue_tail();
            curr->time = TS_TO_TICKS(str->pts);
            /* pts->size should have been zeroed when slot was
               freed */
        }
        else
        {
            /* Add to the one just behind the tail - this may be
             * the head or the previouly added tail - whether or
             * not we'll ever reach this is quite in question
             * since audio always seems to have every packet
             * timestamped */
            curr = audiodesc_queue_tail();
        }

        curr->size += len;

        /* Slide any remainder over to beginning */
        if (audio_queue.ptr > audio_queue.start && audio_queue.used > 0)
        {
            rb->memmove(audio_queue.start, audio_queue.ptr,
                        audio_queue.used);
        }

        /* Splice this packet onto any remainder */
        rb->memcpy(audio_queue.start + audio_queue.used,
                   str->curr_packet, len);

        audio_queue.used += len;
        audio_queue.ptr = audio_queue.start;

        rb->yield();
    }

    return ret;
}

/* Initialise libmad */
static void init_mad(void)
{
    /* init the sbsample buffer */
    frame.sbsample_prev = &sbsample;
    frame.sbsample = &sbsample;

    /* We do this so libmad doesn't try to call codec_calloc(). This needs to
     * be called before mad_stream_init(), mad_frame_inti() and 
     * mad_synth_init(). */
    frame.overlap = &mad_frame_overlap;
    stream.main_data = &mad_main_data;

    /* Call mad initialization. Those will zero the arrays frame.overlap,
     * frame.sbsample and frame.sbsample_prev. Therefore there is no need to 
     * zero them here. */
    mad_stream_init(&stream);
    mad_frame_init(&frame);
    mad_synth_init(&synth);
}

/* Sync audio stream to a particular frame - see main decoder loop for
 * detailed remarks */
static int audio_sync(struct audio_thread_data *td,
                      struct str_sync_data *sd)
{
    int retval = STREAM_MATCH;
    uint32_t sdtime = TS_TO_TICKS(clip_time(&audio_str, sd->time));
    uint32_t time;
    uint32_t duration = 0;
    struct stream *str;
    struct stream tmp_str;
    struct mad_header header;
    struct mad_stream stream;

    if (td->ev.id == STREAM_SYNC)
    {
        /* Actually syncing for playback - use real stream */
        time = 0;
        str = &audio_str;
    }
    else
    {
        /* Probing - use temp stream */
        time = INVALID_TIMESTAMP;
        str = &tmp_str;
        str->id = audio_str.id;
    }        

    str->hdr.pos = sd->sk.pos;
    str->hdr.limit = sd->sk.pos + sd->sk.len;

    mad_stream_init(&stream);
    mad_header_init(&header);

    while (1)
    {
        if (audio_buffer(str, STREAM_PM_RANDOM_ACCESS) == STREAM_DATA_END)
        {
            DEBUGF("audio_sync:STR_DATA_END\n  aqu:%ld swl:%ld swr:%ld\n",
                    (long)audio_queue.used, str->hdr.win_left, str->hdr.win_right);
            if (audio_queue.used <= MAD_BUFFER_GUARD)
                goto sync_data_end;
        }

        stream.error = 0;
        mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);

        if (stream.sync && mad_stream_sync(&stream) < 0)
        {
            DEBUGF(" audio: mad_stream_sync failed\n");
            audio_queue_advance_pos(MAX(audio_queue.curr->size - 1, 1));
            continue;
        }

        stream.sync = 0;

        if (mad_header_decode(&header, &stream) < 0)
        {
            DEBUGF(" audio: mad_header_decode failed:%s\n",
                   mad_stream_errorstr(&stream));
            audio_queue_advance_pos(1);
            continue;
        }

        duration = 32*MAD_NSBSAMPLES(&header);
        time = audio_queue.curr->time;

        DEBUGF(" audio: ft:%u t:%u fe:%u nsamp:%u sampr:%u\n",
               (unsigned)TICKS_TO_TS(time), (unsigned)sd->time,
               (unsigned)TICKS_TO_TS(time + duration),
               (unsigned)duration, header.samplerate);

        audio_queue_advance_pos(stream.this_frame - audio_queue.ptr);

        if (time <= sdtime && sdtime < time + duration)
        {
            DEBUGF(" audio: ft<=t<fe\n");
            retval = STREAM_PERFECT_MATCH;
            break;
        }
        else if (time > sdtime)
        {
            DEBUGF(" audio: ft>t\n");
            break;
        }

        audio_queue_advance_pos(stream.next_frame - audio_queue.ptr);
        audio_queue.curr->time += duration;

        rb->yield();
    }

sync_data_end:
    if (td->ev.id == STREAM_FIND_END_TIME)
    {
        if (time != INVALID_TIMESTAMP)
        {
            time = TICKS_TO_TS(time);
            duration = TICKS_TO_TS(duration);
            sd->time = time + duration;
            retval = STREAM_PERFECT_MATCH;
        }
        else
        {
            retval = STREAM_NOT_FOUND;
        }
    }

    DEBUGF(" audio header: 0x%02X%02X%02X%02X\n",
           (unsigned)audio_queue.ptr[0], (unsigned)audio_queue.ptr[1],
           (unsigned)audio_queue.ptr[2], (unsigned)audio_queue.ptr[3]);

    return retval;
    (void)td;
}

static void audio_thread_msg(struct audio_thread_data *td)
{
    while (1)
    {
        intptr_t reply = 0;

        switch (td->ev.id)
        {
        case STREAM_PLAY:
            td->status = STREAM_PLAYING;

            switch (td->state)
            {
            case TSTATE_INIT:
                td->state = TSTATE_DECODE;
            case TSTATE_DECODE:
            case TSTATE_RENDER_WAIT:
                break;

            case TSTATE_EOS:
                /* At end of stream - no playback possible so fire the
                 * completion event */
                stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
                break;
            }

            break;

        case STREAM_PAUSE:
            td->status = STREAM_PAUSED;
            reply = td->state != TSTATE_EOS;
            break;

        case STREAM_STOP:
            if (td->state == TSTATE_DATA)
                stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);

            td->status = STREAM_STOPPED;
            td->state = TSTATE_EOS;

            reply = true;
            break;            

        case STREAM_RESET:
            if (td->state == TSTATE_DATA)
                stream_clear_notify(&audio_str, DISK_BUF_DATA_NOTIFY);

            td->status = STREAM_STOPPED;
            td->state = TSTATE_INIT;
            td->samplerate = 0;
            td->nchannels = 0;

            init_mad();
            td->mad_errors = 0;

            audio_queue_reset();

            reply = true;
            break;

        case STREAM_NEEDS_SYNC:
            reply = true; /* Audio always needs to */
            break;

        case STREAM_SYNC:
        case STREAM_FIND_END_TIME:
            if (td->state != TSTATE_INIT)
                break;

            reply = audio_sync(td, (struct str_sync_data *)td->ev.data);
            break;

        case DISK_BUF_DATA_NOTIFY:
            /* Our bun is done */
            if (td->state != TSTATE_DATA)
                break;

            td->state = TSTATE_DECODE;
            str_data_notify_received(&audio_str);
            break;

        case STREAM_QUIT:
            /* Time to go - make thread exit */
            td->state = TSTATE_EOS;
            return;
        }

        str_reply_msg(&audio_str, reply);

        if (td->status == STREAM_PLAYING)
        {
            switch (td->state)
            {
            case TSTATE_DECODE:
            case TSTATE_RENDER_WAIT:
                /* These return when in playing state */
                return;
            }
        }

        str_get_msg(&audio_str, &td->ev);
    }
}

static void audio_thread(void)
{
    struct audio_thread_data td;
#ifdef HAVE_PRIORITY_SCHEDULING
    /* Up the priority since the core DSP over-yields internally */
    int old_priority = rb->thread_set_priority(rb->thread_self(),
                                               PRIORITY_PLAYBACK-4);
#endif

    rb->memset(&td, 0, sizeof (td));
    td.status = STREAM_STOPPED;
    td.state = TSTATE_EOS;

    /* We need this here to init the EMAC for Coldfire targets */
    init_mad();

    td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
                                                    CODEC_IDX_AUDIO);
#ifdef HAVE_PITCHSCREEN
    rb->sound_set_pitch(PITCH_SPEED_100);
#endif
    rb->dsp_configure(td.dsp, DSP_RESET, 0);
    rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);

    goto message_wait;

    /* This is the decoding loop. */
    while (1)
    {
        td.state = TSTATE_DECODE;

        /* Check for any pending messages and process them */
        if (str_have_msg(&audio_str))
        {
        message_wait:
            /* Wait for a message to be queued */
            str_get_msg(&audio_str, &td.ev);

        message_process:
            /* Process a message already dequeued */
            audio_thread_msg(&td);

            switch (td.state)
            {
            /* These states are the only ones that should return */
            case TSTATE_DECODE:          goto audio_decode;
            case TSTATE_RENDER_WAIT:     goto render_wait;
            /* Anything else is interpreted as an exit */
            default:
            {
#ifdef HAVE_PRIORITY_SCHEDULING
                rb->thread_set_priority(rb->thread_self(), old_priority);
#endif
                return;
                }
            }
        }

    audio_decode:

        /** Buffering **/
        switch (audio_buffer(&audio_str, STREAM_PM_STREAMING))
        {
        case STREAM_DATA_NOT_READY:
        {
            td.state = TSTATE_DATA;
            goto message_wait;
            } /* STREAM_DATA_NOT_READY: */

        case STREAM_DATA_END:
        {
            if (audio_queue.used > MAD_BUFFER_GUARD)
                break; /* Still have frames to decode */

            /* Used up remainder of compressed audio buffer. Wait for
             * samples on PCM buffer to finish playing. */
            audio_queue_reset();

            while (1)
            {
                if (pcm_output_empty())
                {
                    td.state = TSTATE_EOS;
                    stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);
                    break;
                }

                pcm_output_drain();
                str_get_msg_w_tmo(&audio_str, &td.ev, 1);

                if (td.ev.id != SYS_TIMEOUT)
                    break;
            }

            goto message_wait;
            } /* STREAM_DATA_END: */
        }

        /** Decoding **/
        mad_stream_buffer(&stream, audio_queue.ptr, audio_queue.used);

        int mad_stat = mad_frame_decode(&frame, &stream);

        ssize_t len = stream.next_frame - audio_queue.ptr;

        if (mad_stat != 0)
        {
            DEBUGF("audio: Stream error: %s\n",
                   mad_stream_errorstr(&stream));

            /* If something's goofed - try to perform resync by moving
             * at least one byte at a time */
            audio_queue_advance_pos(MAX(len, 1));

            if (stream.error == MAD_ERROR_BUFLEN)
            {
                /* This makes the codec support partially corrupted files */
                if (++td.mad_errors <= MPA_MAX_FRAME_SIZE)
                {
                    stream.error = 0;
                    rb->yield();
                    continue;
                }
                DEBUGF("audio: Too many errors\n");
            }
            else if (MAD_RECOVERABLE(stream.error))
            {
                /* libmad says it can recover - just keep on decoding */
                rb->yield();
                continue;
            }
            else
            {
                /* Some other unrecoverable error */
                DEBUGF("audio: Unrecoverable error\n");
            }

            /* This is too hard - bail out */
            td.state = TSTATE_EOS;
            td.status = STREAM_ERROR;
            stream_generate_event(&audio_str, STREAM_EV_COMPLETE, 0);

            goto message_wait;
        }

        /* Adjust sizes by the frame size */
        audio_queue_advance_pos(len);
        td.mad_errors = 0; /* Clear errors */

        /* Generate the pcm samples */
        mad_synth_frame(&synth, &frame);

        /** Output **/
        if (frame.header.samplerate != td.samplerate)
        {
            td.samplerate = frame.header.samplerate;
            rb->dsp_configure(td.dsp, DSP_SWITCH_FREQUENCY,
                              td.samplerate);
        }

        if (MAD_NCHANNELS(&frame.header) != td.nchannels)
        {
            td.nchannels = MAD_NCHANNELS(&frame.header);
            rb->dsp_configure(td.dsp, DSP_SET_STEREO_MODE,
                              td.nchannels == 1 ?
                                STEREO_MONO : STEREO_NONINTERLEAVED);
        }

        td.state  = TSTATE_RENDER_WAIT;

        /* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */
    render_wait:
        if (synth.pcm.length > 0)
        {
            const char *src[2] =
                { (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
            int out_count = (synth.pcm.length * CLOCK_RATE
                                + (td.samplerate - 1)) / td.samplerate;
            unsigned char *out_buf;
            ssize_t size = out_count*4;

            /* Wait for required amount of free buffer space */
            while ((out_buf = pcm_output_get_buffer(&size)) == NULL)
            {
                /* Wait one frame */
                int timeout = out_count*HZ / td.samplerate;
                str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1));
                if (td.ev.id != SYS_TIMEOUT)
                    goto message_process;
            }

            out_count = rb->dsp_process(td.dsp, out_buf, src, synth.pcm.length);

            if (out_count <= 0)
                break;

            /* Make this data available to DMA */
            pcm_output_commit_data(out_count*4, audio_queue.curr->time);

            /* As long as we're on this timestamp, the time is just
               incremented by the number of samples */
            audio_queue.curr->time += out_count;
        }

        rb->yield();
    } /* end decoding loop */
}

/* Initializes the audio thread resources and starts the thread */
bool audio_thread_init(void)
{
    /* Initialise the encoded audio buffer and its descriptors */
    audio_queue.start = mpeg_malloc(AUDIOBUF_ALLOC_SIZE,
                                    MPEG_ALLOC_AUDIOBUF);
    if (audio_queue.start == NULL)
        return false;

    /* Start the audio thread */
    audio_str.hdr.q = &audio_str_queue;
    rb->queue_init(audio_str.hdr.q, false);

    /* We steal the codec thread for audio */
    rb->codec_thread_do_callback(audio_thread, &audio_str.thread);

    rb->queue_enable_queue_send(audio_str.hdr.q, &audio_str_queue_send,
                                audio_str.thread);

    /* Wait for thread to initialize */
    str_send_msg(&audio_str, STREAM_NULL, 0);

    return true;
}

/* Stops the audio thread */
void audio_thread_exit(void)
{
    if (audio_str.thread != 0)
    {
        str_post_msg(&audio_str, STREAM_QUIT, 0);
        rb->codec_thread_do_callback(NULL, NULL);
        audio_str.thread = 0;
    }
}